August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Various question - CallerID / Call transfer via REFER and CDR
by Weigel, Stefan 02 Aug '11

02 Aug '11

Hi all, I'm currently stuck with some issues I can't figure out. I'm using FreeSWITCH Version 1.0.head (git-db5f504 2011-07-17 17-00-38 -0400), phone devices are Polycom Soundpoint IP 560. 1. When receiving a call I change some of the incoming Caller-ID names. It's working like a charm. But if I do a call forward (it's a 'Consultative transfer'-Party A dials party C's number and talks privately with party C after the call is answered, and then completes the transfer or hangs up) I only see the party A's number/name on the display C. Is there a possibility to get the original caller-ID/-name displayed on the phone of Party C ? 2. When doing a 'Consultative transfer', I only get one CDR entry using mod_cdr_csv. For example calling from my cellphone 0179 12345678 to extension 615, then doing a call transfer from 615 to 803 I get an entry in CDR with calling_id = 615, destination_number = 803. Is it possible to trigger the creation of a CDR entry ? I tried using process_cdr but only one entry get's logged. In cdr_csv.conf I'm using logging of a-leg, for billing / call statistic. Thanks in advance and best regards Stefan

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Audio issues when FS as PSTN/SBC Gateway for OpenSIPS
by Deon Vermeulen 02 Aug '11

02 Aug '11

Hi List My Current setup is as follows. PSTN -- Sangoma A500 bri -- FS -- OpenSIPS DID: bridges called number to ext@opensips My Phones are located on our office private LAN, while FS and OpenSIPS are both on Public Network with Public IPs. There is NAT between the IP Phones and OpenSIPS/FS. I am using non standard SIP ports as well as not defined any rules for RTP on the firewall. I have a many to one NAT scenario on the Firewall. OpenSIPS is setup to not Proxy any media between the phones and FS. Media flows directly between FS and phones. When I make calls to PSTN, audio is Crystal clear on both sides of the call. When I receive calls from the PSTN I sometimes have audio, sometimes none and most of the time one way (to PSTN). I have done quite a lot of sip trace debugging in fs_cli and I really can't see any problems in the traces. I have also enabled inbound-proxy-media in all sip profiles, just to make sure I don't miss it somewhere in the call flow. I have been digging around in the wiki, forums as well as some archived mails wrt FS and NAT, but I don't 100% understand how FS handles NAT by default in the External.xml profile. I'm also not 100% sure how FS will handle this type of scenario as no Phones are registered to it and it also has to pass SIP and SDP info with OpenSIPS and not directly with the Phones. Any advise and or guidance will really be appreciated. Thank you very much. Kind Regards Deon Vermeulen

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mod_radius_cdr does not work
by fieldpeak 02 Aug '11

02 Aug '11

Dear All, i'm trying to use mod_radius_cdr, and FS successfully loaded it, but when i make a call, it show error below on console, could anyone can help? Thanks in advance. 2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding Freeswitch-Src: 1002 *debug level log on console:* freeswitch@freeswitch> 2011-08-02 14:45:32.546104 [DEBUG] sofia.c:7030 IP 222.128.70.10 Rejected by acl "domains". Falling back to Digest auth. 2011-08-02 14:45:32.546104 [WARNING] sofia_reg.c:1337 SIP auth challenge (INVITE) on sofia profile 'internal' for [1001(a)124.193.106.104] from ip 222.128.70.10 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:7030 IP 222.128.70.10 Rejected by acl "domains". Falling back to Digest auth. 2011-08-02 14:45:32.786095 [NOTICE] switch_channel.c:897 New Channel sofia/internal/1002(a)124.193.106.104 [d8030a1e-552b-4a32-b7f0-175653b65eb7] 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5084 Channel sofia/internal/ 1002(a)124.193.106.104 entering state [received][100] 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5095 Remote SDP: v=0 o=- 186895328 186895344 IN IP4 192.168.200.100 s=eyeBeam c=IN IP4 192.168.200.100 t=0 0 m=audio 6200 RTP/AVP 18 100 6 0 8 3 98 97 5 102 101 a=rtpmap:100 speex/16000 a=rtpmap:98 ilbc/8000 a=rtpmap:97 speex/8000 a=rtpmap:102 l16/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : 21373DA4 8CC12811 192.168.200.100 6200 2011-08-02 14:45:32.786095 [DEBUG] sofia.c:5258 (sofia/internal/ 1002(a)124.193.106.104) State Change CS_NEW -> CS_INIT 2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_INIT 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002(a)124.193.106.104) State INIT 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:85 sofia/internal/ 1002(a)124.193.106.104 SOFIA INIT 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:125 (sofia/internal/ 1002(a)124.193.106.104) State Change CS_INIT -> CS_ROUTING 2011-08-02 14:45:32.786095 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002(a)124.193.106.104) State INIT going to sleep 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_ROUTING 2011-08-02 14:45:32.786095 [DEBUG] switch_channel.c:1821 (sofia/internal/ 1002(a)124.193.106.104) Callstate Change DOWN -> RINGING 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373 (sofia/internal/1002(a)124.193.106.104) State ROUTING 2011-08-02 14:45:32.786095 [DEBUG] mod_sofia.c:148 sofia/internal/ 1002(a)124.193.106.104 SOFIA ROUTING 2011-08-02 14:45:32.786095 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing 2011-08-02 14:45:32.786095 [ERR] mod_radius_cdr.c:242 failed adding Freeswitch-Src: 1002 2011-08-02 14:45:32.786095 [DEBUG] switch_core_state_machine.c:373 (sofia/internal/1002(a)124.193.106.104) State ROUTING going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:855 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] sofia.c:5084 Channel sofia/internal/ 1002(a)124.193.106.104 entering state [terminated][487] 2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2739 (sofia/internal/ 1002(a)124.193.106.104) Callstate Change RINGING -> HANGUP 2011-08-02 14:45:36.185722 [NOTICE] sofia.c:5806 Hangup sofia/internal/ 1002(a)124.193.106.104 [CS_ROUTING] [ORIGINATOR_CANCEL] 2011-08-02 14:45:36.185722 [DEBUG] switch_channel.c:2755 Send signal sofia/internal/1002(a)124.193.106.104 [KILL] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_HANGUP 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/1002(a)124.193.106.104) State HANGUP 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:452 sofia/internal/ 1002(a)124.193.106.104 Overriding SIP cause 487 with 487 from the other leg 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:458 Channel sofia/internal/ 1002(a)124.193.106.104 hanging up, cause: ORIGINATOR_CANCEL 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002(a)124.193.106.104 Standard HANGUP, cause: ORIGINATOR_CANCEL 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:575 (sofia/internal/1002(a)124.193.106.104) State HANGUP going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002(a)124.193.106.104) State Change CS_HANGUP -> CS_REPORTING 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_REPORTING 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/1002(a)124.193.106.104) State REPORTING 2011-08-02 14:45:36.185722 [DEBUG] mod_radius_cdr.c:424 [mod_radius_cdr] Entering my_on_reporting 2011-08-02 14:45:36.185722 [ERR] mod_radius_cdr.c:463 failed adding Freeswitch-Hangupcause: 487 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:635 (sofia/internal/1002(a)124.193.106.104) State REPORTING going to sleep 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1002(a)124.193.106.104) State Change CS_REPORTING -> CS_DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1154 Send signal sofia/internal/1002(a)124.193.106.104 [BREAK] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_session.c:1326 Session 2 (sofia/internal/1002(a)124.193.106.104) Locked, Waiting on external entities 2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1344 Session 2 (sofia/internal/1002(a)124.193.106.104) Ended 2011-08-02 14:45:36.185722 [NOTICE] switch_core_session.c:1346 Close Channel sofia/internal/1002(a)124.193.106.104 [CS_DESTROY] 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:464 (sofia/internal/1002(a)124.193.106.104) Callstate Change HANGUP -> DOWN 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002(a)124.193.106.104) Running State Change CS_DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/1002(a)124.193.106.104) State DESTROY 2011-08-02 14:45:36.185722 [DEBUG] mod_sofia.c:363 sofia/internal/ 1002(a)124.193.106.104 SOFIA DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002(a)124.193.106.104 Standard DESTROY 2011-08-02 14:45:36.185722 [DEBUG] switch_core_state_machine.c:477 (sofia/internal/1002(a)124.193.106.104) State DESTROY going to sleep Regards, Charles

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Last Chance for ClueCon 2011: Get Your Hotel Today
by marketing＠cluecon.com 01 Aug '11

01 Aug '11

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Still having problems making calls to phones behind double nat
by Rob Hutton 01 Aug '11

01 Aug '11

I am still unable to figure this one out and would appreciate any help. I have two phones at a remote location that can call out through the switch and have two way audio, but no one can call into them. The switch returns a 606 User Not registered even though show registrations seems to indicate that they are. I don't see anything in the sip trace to indicate that it actually tried to contact the destination user, but INFO keepalives are going back and forth... Configuration and sofia log are here: http://pastebin.freeswitch.org/16784

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Re: [Freeswitch-users] 3
by Dave Bracken 01 Aug '11

01 Aug '11

Make the right choice.. http://kodaifresh.com/page.php?mgoogleId=69h8

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unable to connect to freeswitch server using x-lite
by Dhairya Vora 01 Aug '11

01 Aug '11

I have installed freeswitch on centos ----------------------------------------------------- [root@localhost ~]# uname -r 2.6.18-238.19.1.el5 ------------------------------------------------------ I have installed x-lite on a different machine in same lan. when i try to connect x-lite to freeswitch with valid id, password, request times out. any idea to solve this? -- Dhairya Vora(newbie)

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